Can't play sound in USB sound card with libasound. (C++)

Question

I'm currently learning to develop using alsa API (libasound). I want to send PCM sound to my usb sound card.

I run this code :

#include <stdio.h>
#include <stdlib.h>
#include <alsa/asoundlib.h>

#include "helloPi.h"

int main(int argc, char *argv[]) {
    int i;
    int err;

    snd_pcm_t *playback_handle;
    snd_pcm_hw_params_t *hw_params;

    if ((err = snd_pcm_open(&playback_handle, "hw:1,0", SND_PCM_STREAM_PLAYBACK,
        0)) < 0) {
        fprintf(stderr, "cannot open audio device %s (%s)\n", argv[1],
            snd_strerror(err));
        exit(1);
    }

    if ((err = snd_pcm_hw_params_malloc(&hw_params)) < 0) {
        fprintf(stderr, "cannot allocate hardware parameter structure (%s)\n",
            snd_strerror(err));
        exit(1);
    }

    if ((err = snd_pcm_hw_params_any(playback_handle, hw_params)) < 0){
        fprintf(stderr, "cannot initialize hardware parameter structure (%s)\n",
            snd_strerror(err));
        exit(1);
    }

    if ((err = snd_pcm_hw_params_set_access(playback_handle, hw_params,
        SND_PCM_ACCESS_RW_INTERLEAVED)) < 0) {
        fprintf(stderr, "cannot set access type (%s)\n", snd_strerror(err));
        exit(1);
    }

    if ((err = snd_pcm_hw_params_set_format(playback_handle, hw_params,
        SND_PCM_FORMAT_S16_LE)) < 0) {
        fprintf(stderr, "cannot set sample format (%s)\n", snd_strerror(err));
        exit(1);
    }

    unsigned int freq = 44100;

    if ((err = snd_pcm_hw_params_set_rate_near(playback_handle, hw_params,
        &freq, 0)) < 0) {
        fprintf(stderr, "cannot set sample rate (%s)\n", snd_strerror(err));
        exit(1);
    }

    if ((err = snd_pcm_hw_params_set_channels(playback_handle, hw_params, 2))
        < 0) {
        fprintf(stderr, "cannot set channel count (%s)\n", snd_strerror(err));
        exit(1);
    }

    if ((err = snd_pcm_hw_params(playback_handle, hw_params)) < 0) {
        fprintf(stderr, "cannot set parameters (%s)\n", snd_strerror(err));
        exit(1);
    }

    snd_pcm_hw_params_free(hw_params);

    if ((err = snd_pcm_prepare(playback_handle)) < 0) {
        fprintf(stderr, "cannot prepare audio interface for use (%s)\n",
            snd_strerror(err));
        exit(1);
    }

    for (i = 0; i < 100; ++i) {
        if ((err = snd_pcm_writei(playback_handle, buf, 2048)) < 0) {
            fprintf(stderr, "write to audio interface failed (%s)\n",
                snd_strerror(err));
            exit(1);
        }
    }

    snd_pcm_close(playback_handle);
    exit(0);
  }

Nota : buf is an array declared in HelloPi.h containing a sound wave.

When I use snd_pcm_open(&playback_handle, "hw:1,0", SND_PCM_STREAM_PLAYBACK,0) with hw:0,0, is work fine with the internal sound card of my laptop. But, if I use hw:1,0, to use the usb sound card, nothing happened (not even an error !).

If I run in a terminal :

aplay -l

I have the following result :

carte 0: PCH [HDA Intel PCH], périphérique 0: ALC283 Analog [ALC283 Analog]
  Sous-périphériques: 1/1
  Sous-périphérique #0: subdevice #0
carte 0: PCH [HDA Intel PCH], périphérique 3: HDMI 0 [HDMI 0]
  Sous-périphériques: 1/1
  Sous-périphérique #0: subdevice #0
carte 1: Set [C-Media USB Headphone Set], périphérique 0: USB Audio [USB Audio]
  Sous-périphériques: 1/1
  Sous-périphérique #0: subdevice #0

Hello

I know that the usb card is OK because I have sound with :

speaker-test -Dhw:1,0 -c2 -twav

I don't know why my code is note producing sound with hw:1,0...

I hope some of you will help me ! Thanks,

Maxime.

Answer 1

I find a solution, I use the code provided in the xample with the asound lib.

/*
 *  This extra small demo sends a random samples to your speakers.
 */

#include <alsa/asoundlib.h>
static char *device = "hw:1,0";         /* playback device */

snd_output_t *output = NULL;
unsigned char buffer[16*1024];              /* some random data */

int main(void)
{
        int err;
        unsigned int i;
        snd_pcm_t *handle;
        snd_pcm_sframes_t frames;

        for (i = 0; i < sizeof(buffer); i++)
                buffer[i] = random() & 0xff;

    if ((err = snd_pcm_open(&handle, device, SND_PCM_STREAM_PLAYBACK, 0)) < 0) {
        printf("Playback1 open error: %s\n", snd_strerror(err));
        exit(EXIT_FAILURE);
    }
    if ((err = snd_pcm_set_params(handle,
                                  SND_PCM_FORMAT_S16,
                                  SND_PCM_ACCESS_RW_INTERLEAVED,
                                  2,
                                  44100,
                                  0,
                                  50000)) < 0) {    /* 0.5sec */
        printf("Playback2 open error: %s\n", snd_strerror(err));
        exit(EXIT_FAILURE);
    }

        for (i = 0; i < 16; i++) {
                frames = snd_pcm_writei(handle, buffer, sizeof(buffer));
                if (frames < 0)
                        frames = snd_pcm_recover(handle, frames, 0);
                if (frames < 0) {
                        printf("snd_pcm_writei failed: %s\n", snd_strerror(frames));
                        break;
                }
                if (frames > 0 && frames < (long)sizeof(buffer))
                        printf("Short write (expected %li, wrote %li)\n", (long)sizeof(buffer), frames);
        }

    snd_pcm_close(handle);
    return 0;
}

It work but I still have to find out why the first code wasn't good.

Bye, Maxime.